Gordon Henderson wrote:[color=blue]
> In article <f2d804$m3m$1$8300dec7@news.demon.co.uk>,
> Steve <user@mail.com> wrote:[color=green]
>> Hi,
>>
>> I tryed to receive phone calls from Sipgate.co.uk
>>
>> I am currently connected to Sipgate.co.uk
>> Now, into my [incoming] calls
>> None of the Sipgate phone received seemed to match this !! :-(
>>
>> --------------------
>> [globals]
>>
>> INCOMING=1009326 ; with phone number : +44 1234 1009326
>>
>> [incoming]
>> Exten => ${INCOMING1},1,answer()
>> Exten => ${INCOMING1},2,Set(TRIES=0)
>> Exten => ${INCOMING1},3,ringing(3)
>> ...........
>> ...........
>>
>> --------------
>>
>> Now, I found on their website :
>> Asterisk PBX
>> IP: sip:sipgate-uk@<My-IP_Address>:5060
>>
>> - Any ideas what's wrong ?
>>
>> I am not sure to understand how to integrate this ?!?[/color]
>
> There is a page on their site which has details, however, try this:
>
> In sip.conf:
>
> ; SIP Trunk to sipgate [sipgate.co.uk]
> register => 1009326:xxxxxxx@sipgate.co.uk/1009326
>
> ; SIP Trunk to sipgate [sipgate.co.uk]
> [sipgate-out]
> context=from-sipgate
> type=friend
> nat=no
> host=sipgate.co.uk
> insecure=port,invite
> username=1009326
> fromuser=1009326
> secret=xxxx
> fromdomain=sipgate.co.uk
> qualify=yes
> disallow=all
> allow=alaw
> allow=g726
> allow=gsm
>
>
> in extensions.conf:
>
> ; Trunk from sipgate [sipgate.co.uk]
> [from-sipgate]
> exten => 1009326,1,Noop(Incoming SIP call from ${CALLERID(all)} @ sipgate calling ${EXTEN})
> exten => 1009326,n,SetCallerID(Via SipGate <${CALLERID(num)}>)
> exten => 1009326,n,Dial(SIP/101)
>
> that last dial line will need to be change to whatever you're using to ring your phone
>
> And somewhere else in extensions.conf, in a contect you can dial-out on:
>
> exten => _70.,1,Noop(Making SIP Call to ${EXTEN:2}@sipgate)
> exten => _70.,n,SetCallerID(1009326)
> exten => _70.,n,Dial(SIP/sipgate-out/${EXTEN:2})
> exten => _70.,n,Hangup()
>
> That's a prefix of 70, then a number. Change it to suit.
>
> Gordon[/color]


Hi Gordon,

Thanks these info.. however, I stil have something wrong with it.
Here is the exact config I have :
I seemed to be connected to the server...
ButNoting is incoming

If I add the line Exten => 0,1,goto(incoming,1009326,1)
and press the 0 => that's OK !!!
SO the script seems to be OK, except that it doesn't work if the phone
call is genereated by Sipgate.

--------------------------

Sip.conf

[general]

port = 5060
bindaddr = 0.0.0.0
allow=all
context=default
srvlookup=no ; could help if Phone using DNS
externip=xx.xx.xx.xx
localnet=192.168.3.0/255.255.255.0

register => 1009326:xxxx@sipgate.co.uk/1009326

[sipgate]
type=friend
secret=xxxxx
username=1009326
fromuser=1009326
fromdomain=sipgate.co.uk
host=sipgate.co.uk
insecure=port,invite
qualify=yes
nat=no
context=incoming
disallow=all
allow=alaw
allow=g726
allow=gsm

EXTENSION.CONF

[general]
;
static=yes
writeprotect=no
Static=yes ; DialPlan can't be modified by Asterisk
clearglobalvars=no
priorityjumping=no
autofallthrough=yes
[globals]

GROUP=SIP/120&SIP/121
INCOMING1=1009326

[incoming]

exten => ${INCOMING1},1,Noop(Incoming SIP call from ${CALLERID(all)} @
sipgate calling ${EXTEN})
exten => ${INCOMING1},2,SetCallerID(From SipGate <${CALLERID(num)}>)
Exten => ${INCOMING1},2,Set(TRIES=0)
Exten => ${INCOMING1},3,ringing(3)
Exten => ${INCOMING1},4,background(priv-introsaved)
Exten => ${INCOMING1},5,background(queue-thankyou)
Exten => ${INCOMING1},n,Dial(SIP/121,10,r)
Exten => ${INCOMING1},n,playback(vm-nobodyavail)
Exten => ${INCOMING1},n,Dial(${GROUP},10,mr)
Exten => ${INCOMING1},n,playback(tt-allbusy)
Exten => ${INCOMING1},n,wait(2)
Exten => ${INCOMING1},n,goto(${INCOMING1},7)
;Exten => ${INCOMING1},n,voicemail(u300@default)
Exten => ${INCOMING1},n,congestion